[SR-Users] No audio when calling from SIP to WEBRTC

[Daniel-Constantin Mierla]

Hello,

I see this was discussed further on rtpengine issue tracker. Did using a
newer version of rtpengine made it work?

The typical hint I have is to look at javascript console in the browser,
there should be logs printed when some dtls negotiation fails.

Cheers,
Daniel

On 05.05.20 02:21, Chirag Desai wrote:
>
> Hi all,
>
> I have configured Kamailio for WebSockets following this guide as an
> example: https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg
>
> With sip.js and jssip I'm able to initiate a call from WebRTC to SIP
> and establish a call successfully.
>
> The issue arises when I try to receive a call from a SIP device. In
> this case the call establishes but there is no audio in either direction.
>
> I *think* the issue is with RTP Engine and I've raised a bug there,
> but I'm not sure why it is
> misbehaving https://github.com/sipwise/rtpengine/issues/983. There are
> some logs from RTP engine posted here.
>
> The sip device communicates with Kamailio over UDP / RTP, nothing is
> encrypted.
>
> I would appreciate any guidance.
>
> Thanks in advance,
>
> C
>
>
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> Kamailio (SER) - Users Mailing List
> sr-users at lists.kamailio.org
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-- 
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Funding: https://www.paypal.me/dcmierla

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